3.3.9.3 (January 26, 2015) - Update ReleaseFor PBX- Fixed the bug where timer and schedule did not work correctly when the system time switches to and from summer time- Fixed a rare bug where media server get disconnected- Fixed the bug where failed to record calls after a call transfer- Fixed the bug where the configuration settings of RFC2833 in [Options] did not set correctly- Fixed minor bugs

3.3.7.4 (September 19, 2014) - Update ReleaseFor PBX- Ability to accept 60msec of codec G.729 - Fixed the bug where handling 40msec of codec G.729.- Fixed the bug where codec handling of incoming sessions- Fixed the bug where call parking placed with the keypad command did not work properly when pick up was tried before the call disconnected by the person who was placing the parked call- Fixed minor bugs

For SIP Server- Fixed the bug where the Radius Accounting fails under heavy load- Fixed the bug where the TCP handling occasionally freezes when the TCP connection is disconnected unexpectedly- Fixed minor bugs

3.3.5.8 (July 1, 2014) - Update ReleaseFor PBX- Fixed the bug that call park placed using dial keypad command does not function correctly when picked up before hanging up- Fixed the bug in call conference feature (v3.3.5.4 only)- Changed the default setting length to determine deletion of the recorded file to 2 seconds from 0.2 seconds- Fixed the bug that connection was allowed to use Brekeke PAL WebSocket when incorrect password was used- Fixed minor bugs

For SIP Server- Fixed the bug that the details of Dial Plan History cannot be displayed- Fixed the bug where re-INVITE after 407 might not be resend correctly- Fixed the bug where keep-alive SIP packet doesn't have the valid SIP-URI in To header- Fixed the bug where the SIP Server didn't parse OS's network configuration file correctly in certain Linux distributions

3.3.5.4 (April 28, 2014) - Update ReleaseFor PBX- Fixed the bug that occurs rarely, which the same user name cannot be created after a deleting a user- Fixed the bug that occurs rarely, which call transfer failed when it is transferred from an auto attendant- Fixed the bug that user class was not applied correctly- Fixed minor bugs

3.3.4.4 (March 3, 2014) - Standard ReleaseFor PBX- Added option to specify FROM as conference number when conference is called automatically.- Fixed the bug that multiple callee are sporadically connected when simultaneous ring is used.- Fixed the bug that processing SDP on 18x- Ability to set result in script module of IVR Flow- Fiexed minor bugs

For SIP Server- Support certificate chaining for TLS- Support PEM formatted certificate and key files for TLS- Fixed the bug where the TCP/TLS connection might not accept REGISTRER if it indicates "Contact: *".- Fixed the bug where the [Active Sessions] page didn't show sessions made by the previous day. - Fixed the bug where the [Active Sessions] page might not show sessions correctly if there are SUBSCRIBE sessions.- Fixed the bug where the [Remote Address Pattern] didn't work if it is for the exiting interface address.

For SIP Server- Fixed the bug where the SIP proxy kept sending 407 to SUBSCRIBE- Fixed the bug where the Mirroring function didn't handle TLScorrectly after failover happened.- Fixed the bug where the Mirroring function didn't work when both TLS and TCP are used simultaneously. - Added the [Do not Block Local IP Address] in the Block List function for excluding SIP packets sent from local network- Fixed minor bugs

3.1.9.0 (July 15, 2013) - Update ReleaseFor PBX- Fixed the bug where RTP relay did not work when PBX tries to use the port that is already in used by other application (Windows OS)- Fixed the bug where call recording did not work when PBX tries to use the port that is already in used by other application (Windows OS)- Fixed the bug in Japanese language display where user type can be changed even when one is logged in as a user

For SIP Server- Fixed the bug where RTP relay did not work when SIP server tries to use the port that is already in used by other application (Windows OS)- Fixed the bug where SIP server reject CANCEL with a response, "481 Call Leg/Transaction Does Not Exist", when CANCEL was received before INVITE session was made.

3.1.8.2 (June 04, 2013) - Update ReleaseFor PBX- Fixed the bug where KEE_COPY property didn't work in RecordingFileHttpUploader plug-in- Fixed the bug where ARS settings were not saved correctly when Backup/Restore is used- Fixed the bug where restriction of IP addresses didn't work with 3PCC- Fixed minor bugs

For SIP Server- Fixed the bug where value set at "Timeout" and "Interval" were not applied at Heatbeat setting- Fixed the bug where the TLS listener didn't handle the case correctly when connection timeout happen during the handshaking process- Fixed minor bugs

3.1.7.8 (March 26, 2013) - Update ReleaseFor SIP Server- Fixed the bug where SIP server could not handle RTP relay when the size of the RTP packet is large

3.1.7.0 (March 1, 2013) - Update ReleaseFor PBX- Fixed the bug where RTP session timeout did not work under certain conditions- Fixed the bug where timeout for transfer from IVR was not handled accurately- Fixed the bug where the log parameters of recording file in the property file was incorrectly entered- Improved call recording performance- Fixed minor bugsFor SIP Server- Fixed the bug where the SIP proxy did not handle CANCEL correctly when the destination requests INVITE authentication- Fixed the bug where RTP session timeout did not work under certain conditions- Fixed minor bugs

3.1.5.8 (January 24, 2013) - Update ReleaseFor PBX- Fixed the bug where system deletes recorded file in next day when [Conversation recording file in voicemail inbox] is set to "no".- Fixed the bug where call cannot be made when display name of "To" header includes "&" character.- Fixed the bug where secondary server settings are not reflected when ARS route has been deleted on the primary server.- Fixed minor bugsFor SIP Server- Fixed the bug where the SIP proxy might not release session resources when a subsequent SIP request contains an invalid From-tag.

3.1.4.4 (December 14, 2012) - Standard ReleaseFor PBX- Improved call log feature and added link to download call recording files from the call log view.- Added web service interfaces for modifying the ARS route values.- Fixed some bugs in the SCA feature- Fixed the bug where the ARS session counter occasionally did not work correctly.- Made some improvements on ARS/DID pages- Fixed the early media was not relayed correctly with 3PCC calls.- Improved user access.- Fixed minor bugsFor SIP Server- Fixed the bug where the memory leak occurs when there are heavy traffic.- Fixed the bug where the Advanced Edition might not release TCP/TLS resources when a phone disconnects TCP/TLS connection in each SUBSCRIBE request.- Fixed the bug where the SIP server may not automatically enable RTP-relay in the default settings when content-type contain a capital letter.- Fixed the bug where the SIP server may not parse some formats of body which sent from TCP/TLS.- Added GUI settings for authenticating SUBSCRIBE and MESSAGE requests.- Fixed minor bugs

3.1.3.0 (November 9, 2012) - Beta ReleaseFor PBX- Added an option to terminate to conference call when the host leave the line.- Made some improvement on ARS/DID pages- Fixed the bug where callback did not work when the caller's [Max Inbound Session] = 1.For SIP Server- Improved performance when executing Session Plug-in with TCP/TLS connection.- Fixed the bug where Record-Route headers in response packets occasionally replaced incorrectly when TLS is used.- Fixed the bug where $transport, $ifsrc and $ifdst values were cleared if multiple rules were matched.- Fixed the bug where Mirroring Virtual IP address was not applied as an interface address.- Fixed the bug where TCP/TLS connecter causes a freeze.- Fixed minor bugs

3.0.7.0 (July 20, 2012) - Update ReleaseFor PBX- Fixed the bug where tenant settings are failed to be applied without restart- Fixed the bug where schedule was not applied when the end year field is left empty.- Improved security for logging in the product administrative tool.- Improved shutdown process- Fixed minor bugsFor SIP Server- Fixed the bug where the Subscribe Session does not accept NOTIFY at times.- Fixed minor bugs

3.0.6.3 (June 31, 2012) - Update ReleaseFor PBX- Fixed the bug where Call Forwarding with 3xx response from callee did not work.- Improve voice quality when there are multiple G.729 sessions used.- Improve the upgrade method for version 2 to version 3.- Fixed the bug where the license randomly resets when the program restarted.- Fixed minor bugsFor SIP Server- Change the default value of IPv6 to "off"- Change the default value of TLS-handling to "off"- Change the default value of Check Request-URI's validity to "off"- Fixed a bug that the hide loopback didn't work.- Fixed minor bugs

3.0.5.5 (May 23, 2012) - Standard ReleaseFor PBX- Fixed a bug where call recording did not work under certain operations- Fixed a bug where auto monitoring did not work under certain operations- Fixed a bug where input rule in IVR designer did not work- Fixed a bug where 3PCC did not work with IVR- Fixed a bug where date condition did not work correctly- Fixed minor bugsFor SIP Server- Added Failover plugin- Increased the default max number of shared SIP session per thread to 50- Fixed the bug where DNS SRV for load-balancing did not work under certain conditions- Fixed minor bugs

2.4.9.0 (May 5, 2011) - Update ReleaseFor PBX- Fixed the bug where KEEP_COPY parameter in RecordingFileHttpUploader of Audio File Plug-in did not work. - Fixed the bug where failed to release memory during SUBSCRIBE session- Fixed the bug where Brekeke PAL did not retrieve correct information when features like call pick up is used- Fixed the bug where authorized settings for tenant administrator in Multi-Tenant Edition- Fixed the bug where SCA's information did not correctly removed- Add option to include Phone ID in log database

2.4.8.6 (March 25, 2011) - Update ReleaseFor PBX- Optimized performance for BLF (Busy Lamp Field). - Fixed the bug where Brekeke PAL had problem with releasing memory when it is used with a particular SIP UA.

For SIP Server- Fixed the bug where Import Users was not working correctly. - Support Redirection by Failover plugin. - Fixed the bug where Failover plugin with B2B mode didn't clear resending packets. - Added session plug-in for modifying SDP content. - Fixed the bug where the proxy could not handle CSeq which Cisco BTS 10200 Softswitch generates. - Fixed the bug where the registrar may not point reachable port number if a user registers over TCP. - Fixed the bug where the webget will hang if a destination web server doesn't respond. - Fixed the bug where a destination of resending "200 OK" responses may differ from the first packet.

For SIP Server - Fixed the bug where $request definition in Deploy Patterns did not work. - Fixed the bug where Upper/Thru registration did not include some headers and parameters. - Included Failover-Plug-in - Fixed minor bugs

For SIP Server- Fixed the bug where Unregister button failed to respond in multi-domain mode - Fixed the bug where "Disconnected by System" recorded at call log when the session is not disconnected by system. - Fixed minor bugs

2.3.8.2 (September 24, 2009) - Update Release For PBX - Fixed a bug where a session counter of ARS did not work correctly. - Fixed a bug where ARS failover did not work correctly when [Disable on failure]="This group". - Fixed minor bugs

For SIP Server- Fixed the bug where agents could not register from Admin GUI.- Display **** for Password at Heartbeat page.- Fixed the bug where restore did not work when there is no Heartbeat settings.- Fixed minor bugs.

For SIP Server- Default values of authentication for REGISTER and INVITE are "on"

2.2.7.7 (March 9, 2009) - Version Update for PBX- No changes.

for SIP Server- Fixed a bug where the Brekeke SIP Server Admintool may lose some settings under certain operations.

2.2.7.6 (March 6, 2009) - Version Update for PBX- Fixed the bug where there is no audio when call has been picked up with RTP relay set to "off".- Added default regular expression (^.+$) at user field under ARS Pattern OUT- Fixed the bug related to 302 response- Added restriction where calls will be rejected when the caller/callee is neither a PBX user nor going through ARS route settings - Fixed the bug related to RTP relay

For SIP Server- Fixed some bugs in the Mirroring-Mode. - Fixed the bug where the Thread-Sharing did not handle multiple resends correctly. - Fixed a minor bug in the NAT-detection.

2.2.5.8 (November 5, 2008) - Official Release for PBX -Fixed rare Access Violation that happens when PBX handles many concurrent call-recording sessions. -Fixed the bug where calls could not be disconnected when a caller hangs up the call using auto attendant. -Fixed the bug where [Codec Priority] at the Media Server settings was not applied. -Fixed minor bugs.

for SIP Server - Fixed some bugs in the Mirroring-Mode. - Fixed the bug where the SIP server didn't accept the setting of "RTP relay (UA on this machine)". - Improved the NAT detection.

for SIP Server- Fixed the bug where NOTIFY/OPTIONS/MESSAGE messages consume system memory and cause a system to go down. - More stable TCP connectivity - Support switching of transport in mid-session. - Follow the RFC more tightly. - DialPlan: Response Header Definition - Send 407 before sending 404. In previous version, the SIP Server sends 404 when the not-found happens even if the authentication is enabled. From this version, the SIP Server authenticates the request before it sends 404.

for SIP Server - Advanced Edition was added to the Product Line (The Edition Comparison is at http://www.brekeke.com/products/products_sip_2.php) - Added ability to modify SDP's addresses using the DialPlan - Fixed minor bugs

2.1.2.2 (Aug 23, 2007) for PBX: - Fixed the bug that the call entered in to call queue was disconnected occasionally. - Fixed the bug that a PBX user (callee) was called occasionally even [Ringer Time] was set to 0. - Fixed the bug that some dll did not work correctlly during software update. - Fixed minor bugs

2.0.7.0 (Feb 14, 2007) for PBX: - Call Forwarding methods when [Round Robin] is set has been updated and improved. - Removed version display at SIP UAs. - When voicemail is checked at Call forwarding settings, the entry field is left blank. - Default length for [Ringing Timeout (ms)] under Option menu is changed to 4 minutes from 2 minutes. - MARK flag in RTP packets is set to "off". - Fixed bug with cancelling calls when call was terminated simultaneously when session was connected. - Fixed bug for when a UA rings for a second while the [Ringer time (sec)] setting was "0". - Fixed bug for returning "481" against "PRACK" when it has unexpected RSEQ. - Fixed bug where the timestamp for RTP packets becomes out of order when a blind transfer is initiated from user agent/SIP phone.

2.0.4.4 Beta (Dec 13, 2006) for PBX: - Fixed a problem with the mediaserver from 2.0.4.1 beta - Fixed the bug related to a memory leak. - Changed the behavior for returning to a conversation by dialing "**" after a call is dropped while the call is on hold.

for SIP Server: - Supports UPnP for detecting a router and its global IP address, and making the port mapping. - Ability to specify the pattern of additional external IP addresses. - Ability to search the string from the SDP by using the DialPlan.

2.0.1.6 Beta (Sept 11, 2006) for PBX: - Included a new feature called "Automatic Monitoring" it allows the designated extension to monitor the calls made from/to the particular extension . - Minor bug fixes

1.5.3.0 (July 31, 2006) - Fixed the bug that stops pbx running when logging off Windows - Corrected the problem of attended transfer for some SIP phones. - Fixed the bug that drops the call, after transferred, when it reaches the maximum value set in the field "Conversation recording length (sec)".

1.5.2.0 (April 10, 2006) - Fixed the RTP relay problem, caller can not hear callee, when it is set to on(G.711 u only) after the ARS fail over - Fixed the intermittent problem of RTP relay - PBX now can handle Alaw 30ms packet correctly.

1.5.1.3 (Dec 16, 2005) -Fixed a bug that Call Park didn't work when Call Park number range (Park number (min) & Park nmber (max)) set in Option menu is not 2 digits.

1.5.1.1 Beta (Nov 30, 2005) - Added a feature that a user can call multiple users at a time for starting a conference by dialing +*+*+... - Fixed a bug that listening-only mode didn't work even you dialed the prefix 7* to join in the conversation. - Fixed minor bugs

1.5.0.8 Beta (Nov 17, 2005) - Add "Call Status" page in the OnDO PBX Admintool to show status of active calls. - Fixed a bug that ARS fail over does not work when making calls to conference members.

1.4.5.0 Beta (Aug 19, 2005) - Added an option for SMTP authentication = on or off. - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when no response is arrived at OnDO PBX. - From this version, OnDO PBX will retry sending REGISTER request to other SIP Server (such as ITSPs) after a fixed time even Authentication error has occurred. By default, the fixed time = 1800000 milliseconds (30 minutes)

1.4.4.5 (Oct 18, 2005) - Fixed the problem that voice wasn't transmitted from OnDO PBX when RTP relay = on at OnDO PBX.(It happened on very rare occasions) - Fixed the problem that user can not hear audio after transfering a call using #9 when Call Recording was enabled. (It happened on very rare occasions) - Fixed the problem that RTP relay stopped working after PBX receiving re-INVITE from UA. (It happened on very rare occasions)

1.4.4.3 (Aug 19, 2005) - Added an option for SMTP authentication = on or off. - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when no response arrived at OnDO PBX. - From this version, OnDO PBX will retry sending REGISTER request to other SIP Server (such as ITSPs) after a fixed time even Authentication error has occurred. By default, the fixed time = 1800000 milliseconds (30 minutes)

1.4.4.2 Beta (July 20, 2005) - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when the Realm for INVITE is different from the Realm for REGISTER. - Added a Date header in the Email for a Voicemail Email notification (Fixed the compatibility issue with SMTP Server). - WAV sound format of Voicemail file that you can download from OnDO PBX admintool or that is attached in the Email notification is changed from PCM to u-law.

1.4.4.0 (Jun 27, 2005) - Supports qop="auth,auth-int" for some ITSPs. - Fixed a bug that OnDO PBX didn't send ACK when 200 OK from OnDO PBX and CANCEL from a UA were sent at the same time. - Fixed a problem that happened only on a 64-bit Solaris. - Fixed a problem that some sessions were remained after calls for some cases. - Fixed minor bugs

1.3.1.9 (Dec 15, 2004) - A bug that a display name in FROM header is deleted when a call goes through Auto Attendant, was fixed. - A problem that no voice is transmitted after RTP session timeout when [RTP relay] = on, was fixed.

3.3.9.3 (January 26, 2015) - Update ReleaseFor PBX- Fixed the bug where timer and schedule did not work correctly when the system time switches to and from summer time- Fixed a rare bug where media server get disconnected- Fixed the bug where failed to record calls after a call transfer- Fixed the bug where the configuration settings of RFC2833 in [Options] did not set correctly- Fixed minor bugs

3.3.7.4 (September 19, 2014) - Update ReleaseFor PBX- Ability to accept 60msec of codec G.729 - Fixed the bug where handling 40msec of codec G.729.- Fixed the bug where codec handling of incoming sessions- Fixed the bug where call parking placed with the keypad command did not work properly when pick up was tried before the call disconnected by the person who was placing the parked call- Fixed minor bugs

For SIP Server- Fixed the bug where the Radius Accounting fails under heavy load- Fixed the bug where the TCP handling occasionally freezes when the TCP connection is disconnected unexpectedly- Fixed minor bugs

3.3.5.8 (July 1, 2014) - Update ReleaseFor PBX- Fixed the bug that call park placed using dial keypad command does not function correctly when picked up before hanging up- Fixed the bug in call conference feature (v3.3.5.4 only)- Changed the default setting length to determine deletion of the recorded file to 2 seconds from 0.2 seconds- Fixed the bug that connection was allowed to use Brekeke PAL WebSocket when incorrect password was used- Fixed minor bugs

For SIP Server- Fixed the bug that the details of Dial Plan History cannot be displayed- Fixed the bug where re-INVITE after 407 might not be resend correctly- Fixed the bug where keep-alive SIP packet doesn't have the valid SIP-URI in To header- Fixed the bug where the SIP Server didn't parse OS's network configuration file correctly in certain Linux distributions

3.3.5.4 (April 28, 2014) - Update ReleaseFor PBX- Fixed the bug that occurs rarely, which the same user name cannot be created after a deleting a user- Fixed the bug that occurs rarely, which call transfer failed when it is transferred from an auto attendant- Fixed the bug that user class was not applied correctly- Fixed minor bugs

3.3.4.4 (March 3, 2014) - Standard ReleaseFor PBX- Added option to specify FROM as conference number when conference is called automatically.- Fixed the bug that multiple callee are sporadically connected when simultaneous ring is used.- Fixed the bug that processing SDP on 18x- Ability to set result in script module of IVR Flow- Fiexed minor bugs

For SIP Server- Support certificate chaining for TLS- Support PEM formatted certificate and key files for TLS- Fixed the bug where the TCP/TLS connection might not accept REGISTRER if it indicates "Contact: *".- Fixed the bug where the [Active Sessions] page didn't show sessions made by the previous day. - Fixed the bug where the [Active Sessions] page might not show sessions correctly if there are SUBSCRIBE sessions.- Fixed the bug where the [Remote Address Pattern] didn't work if it is for the exiting interface address.

For SIP Server- Fixed the bug where the SIP proxy kept sending 407 to SUBSCRIBE- Fixed the bug where the Mirroring function didn't handle TLScorrectly after failover happened.- Fixed the bug where the Mirroring function didn't work when both TLS and TCP are used simultaneously. - Added the [Do not Block Local IP Address] in the Block List function for excluding SIP packets sent from local network- Fixed minor bugs

3.1.9.0 (July 15, 2013) - Update ReleaseFor PBX- Fixed the bug where RTP relay did not work when PBX tries to use the port that is already in used by other application (Windows OS)- Fixed the bug where call recording did not work when PBX tries to use the port that is already in used by other application (Windows OS)- Fixed the bug in Japanese language display where user type can be changed even when one is logged in as a user

For SIP Server- Fixed the bug where RTP relay did not work when SIP server tries to use the port that is already in used by other application (Windows OS)- Fixed the bug where SIP server reject CANCEL with a response, "481 Call Leg/Transaction Does Not Exist", when CANCEL was received before INVITE session was made.

3.1.8.2 (June 04, 2013) - Update ReleaseFor PBX- Fixed the bug where KEE_COPY property didn't work in RecordingFileHttpUploader plug-in- Fixed the bug where ARS settings were not saved correctly when Backup/Restore is used- Fixed the bug where restriction of IP addresses didn't work with 3PCC- Fixed minor bugs

For SIP Server- Fixed the bug where value set at "Timeout" and "Interval" were not applied at Heatbeat setting- Fixed the bug where the TLS listener didn't handle the case correctly when connection timeout happen during the handshaking process- Fixed minor bugs

3.1.7.8 (March 26, 2013) - Update ReleaseFor SIP Server- Fixed the bug where SIP server could not handle RTP relay when the size of the RTP packet is large

3.1.7.0 (March 1, 2013) - Update ReleaseFor PBX- Fixed the bug where RTP session timeout did not work under certain conditions- Fixed the bug where timeout for transfer from IVR was not handled accurately- Fixed the bug where the log parameters of recording file in the property file was incorrectly entered- Improved call recording performance- Fixed minor bugsFor SIP Server- Fixed the bug where the SIP proxy did not handle CANCEL correctly when the destination requests INVITE authentication- Fixed the bug where RTP session timeout did not work under certain conditions- Fixed minor bugs

3.1.5.8 (January 24, 2013) - Update ReleaseFor PBX- Fixed the bug where system deletes recorded file in next day when [Conversation recording file in voicemail inbox] is set to "no".- Fixed the bug where call cannot be made when display name of "To" header includes "&" character.- Fixed the bug where secondary server settings are not reflected when ARS route has been deleted on the primary server.- Fixed minor bugsFor SIP Server- Fixed the bug where the SIP proxy might not release session resources when a subsequent SIP request contains an invalid From-tag.

3.1.4.4 (December 14, 2012) - Standard ReleaseFor PBX- Improved call log feature and added link to download call recording files from the call log view.- Added web service interfaces for modifying the ARS route values.- Fixed some bugs in the SCA feature- Fixed the bug where the ARS session counter occasionally did not work correctly.- Made some improvements on ARS/DID pages- Fixed the early media was not relayed correctly with 3PCC calls.- Improved user access.- Fixed minor bugsFor SIP Server- Fixed the bug where the memory leak occurs when there are heavy traffic.- Fixed the bug where the Advanced Edition might not release TCP/TLS resources when a phone disconnects TCP/TLS connection in each SUBSCRIBE request.- Fixed the bug where the SIP server may not automatically enable RTP-relay in the default settings when content-type contain a capital letter.- Fixed the bug where the SIP server may not parse some formats of body which sent from TCP/TLS.- Added GUI settings for authenticating SUBSCRIBE and MESSAGE requests.- Fixed minor bugs

3.1.3.0 (November 9, 2012) - Beta ReleaseFor PBX- Added an option to terminate to conference call when the host leave the line.- Made some improvement on ARS/DID pages- Fixed the bug where callback did not work when the caller's [Max Inbound Session] = 1.For SIP Server- Improved performance when executing Session Plug-in with TCP/TLS connection.- Fixed the bug where Record-Route headers in response packets occasionally replaced incorrectly when TLS is used.- Fixed the bug where $transport, $ifsrc and $ifdst values were cleared if multiple rules were matched.- Fixed the bug where Mirroring Virtual IP address was not applied as an interface address.- Fixed the bug where TCP/TLS connecter causes a freeze.- Fixed minor bugs

3.0.7.0 (July 20, 2012) - Update ReleaseFor PBX- Fixed the bug where tenant settings are failed to be applied without restart- Fixed the bug where schedule was not applied when the end year field is left empty.- Improved security for logging in the product administrative tool.- Improved shutdown process- Fixed minor bugsFor SIP Server- Fixed the bug where the Subscribe Session does not accept NOTIFY at times.- Fixed minor bugs

3.0.6.3 (June 31, 2012) - Update ReleaseFor PBX- Fixed the bug where Call Forwarding with 3xx response from callee did not work.- Improve voice quality when there are multiple G.729 sessions used.- Improve the upgrade method for version 2 to version 3.- Fixed the bug where the license randomly resets when the program restarted.- Fixed minor bugsFor SIP Server- Change the default value of IPv6 to "off"- Change the default value of TLS-handling to "off"- Change the default value of Check Request-URI's validity to "off"- Fixed a bug that the hide loopback didn't work.- Fixed minor bugs

3.0.5.5 (May 23, 2012) - Standard ReleaseFor PBX- Fixed a bug where call recording did not work under certain operations- Fixed a bug where auto monitoring did not work under certain operations- Fixed a bug where input rule in IVR designer did not work- Fixed a bug where 3PCC did not work with IVR- Fixed a bug where date condition did not work correctly- Fixed minor bugsFor SIP Server- Added Failover plugin- Increased the default max number of shared SIP session per thread to 50- Fixed the bug where DNS SRV for load-balancing did not work under certain conditions- Fixed minor bugs

2.4.9.0 (May 5, 2011) - Update ReleaseFor PBX- Fixed the bug where KEEP_COPY parameter in RecordingFileHttpUploader of Audio File Plug-in did not work. - Fixed the bug where failed to release memory during SUBSCRIBE session- Fixed the bug where Brekeke PAL did not retrieve correct information when features like call pick up is used- Fixed the bug where authorized settings for tenant administrator in Multi-Tenant Edition- Fixed the bug where SCA's information did not correctly removed- Add option to include Phone ID in log database

2.4.8.6 (March 25, 2011) - Update ReleaseFor PBX- Optimized performance for BLF (Busy Lamp Field). - Fixed the bug where Brekeke PAL had problem with releasing memory when it is used with a particular SIP UA.

For SIP Server- Fixed the bug where Import Users was not working correctly. - Support Redirection by Failover plugin. - Fixed the bug where Failover plugin with B2B mode didn't clear resending packets. - Added session plug-in for modifying SDP content. - Fixed the bug where the proxy could not handle CSeq which Cisco BTS 10200 Softswitch generates. - Fixed the bug where the registrar may not point reachable port number if a user registers over TCP. - Fixed the bug where the webget will hang if a destination web server doesn't respond. - Fixed the bug where a destination of resending "200 OK" responses may differ from the first packet.

For SIP Server - Fixed the bug where $request definition in Deploy Patterns did not work. - Fixed the bug where Upper/Thru registration did not include some headers and parameters. - Included Failover-Plug-in - Fixed minor bugs

For SIP Server- Fixed the bug where Unregister button failed to respond in multi-domain mode - Fixed the bug where "Disconnected by System" recorded at call log when the session is not disconnected by system. - Fixed minor bugs

2.3.8.2 (September 24, 2009) - Update Release For PBX - Fixed a bug where a session counter of ARS did not work correctly. - Fixed a bug where ARS failover did not work correctly when [Disable on failure]="This group". - Fixed minor bugs

For SIP Server- Fixed the bug where agents could not register from Admin GUI.- Display **** for Password at Heartbeat page.- Fixed the bug where restore did not work when there is no Heartbeat settings.- Fixed minor bugs.

For SIP Server- Default values of authentication for REGISTER and INVITE are "on"

2.2.7.7 (March 9, 2009) - Version Update for PBX- No changes.

for SIP Server- Fixed a bug where the Brekeke SIP Server Admintool may lose some settings under certain operations.

2.2.7.6 (March 6, 2009) - Version Update for PBX- Fixed the bug where there is no audio when call has been picked up with RTP relay set to "off".- Added default regular expression (^.+$) at user field under ARS Pattern OUT- Fixed the bug related to 302 response- Added restriction where calls will be rejected when the caller/callee is neither a PBX user nor going through ARS route settings - Fixed the bug related to RTP relay

For SIP Server- Fixed some bugs in the Mirroring-Mode. - Fixed the bug where the Thread-Sharing did not handle multiple resends correctly. - Fixed a minor bug in the NAT-detection.

2.2.5.8 (November 5, 2008) - Official Release for PBX -Fixed rare Access Violation that happens when PBX handles many concurrent call-recording sessions. -Fixed the bug where calls could not be disconnected when a caller hangs up the call using auto attendant. -Fixed the bug where [Codec Priority] at the Media Server settings was not applied. -Fixed minor bugs.

for SIP Server - Fixed some bugs in the Mirroring-Mode. - Fixed the bug where the SIP server didn't accept the setting of "RTP relay (UA on this machine)". - Improved the NAT detection.

for SIP Server- Fixed the bug where NOTIFY/OPTIONS/MESSAGE messages consume system memory and cause a system to go down. - More stable TCP connectivity - Support switching of transport in mid-session. - Follow the RFC more tightly. - DialPlan: Response Header Definition - Send 407 before sending 404. In previous version, the SIP Server sends 404 when the not-found happens even if the authentication is enabled. From this version, the SIP Server authenticates the request before it sends 404.

for SIP Server - Advanced Edition was added to the Product Line (The Edition Comparison is at http://www.brekeke.com/products/products_sip_2.php) - Added ability to modify SDP's addresses using the DialPlan - Fixed minor bugs

2.1.2.2 (Aug 23, 2007) for PBX: - Fixed the bug that the call entered in to call queue was disconnected occasionally. - Fixed the bug that a PBX user (callee) was called occasionally even [Ringer Time] was set to 0. - Fixed the bug that some dll did not work correctlly during software update. - Fixed minor bugs

2.0.7.0 (Feb 14, 2007) for PBX: - Call Forwarding methods when [Round Robin] is set has been updated and improved. - Removed version display at SIP UAs. - When voicemail is checked at Call forwarding settings, the entry field is left blank. - Default length for [Ringing Timeout (ms)] under Option menu is changed to 4 minutes from 2 minutes. - MARK flag in RTP packets is set to "off". - Fixed bug with cancelling calls when call was terminated simultaneously when session was connected. - Fixed bug for when a UA rings for a second while the [Ringer time (sec)] setting was "0". - Fixed bug for returning "481" against "PRACK" when it has unexpected RSEQ. - Fixed bug where the timestamp for RTP packets becomes out of order when a blind transfer is initiated from user agent/SIP phone.

2.0.4.4 Beta (Dec 13, 2006) for PBX: - Fixed a problem with the mediaserver from 2.0.4.1 beta - Fixed the bug related to a memory leak. - Changed the behavior for returning to a conversation by dialing "**" after a call is dropped while the call is on hold.

for SIP Server: - Supports UPnP for detecting a router and its global IP address, and making the port mapping. - Ability to specify the pattern of additional external IP addresses. - Ability to search the string from the SDP by using the DialPlan.

2.0.1.6 Beta (Sept 11, 2006) for PBX: - Included a new feature called "Automatic Monitoring" it allows the designated extension to monitor the calls made from/to the particular extension . - Minor bug fixes

1.5.3.0 (July 31, 2006) - Fixed the bug that stops pbx running when logging off Windows - Corrected the problem of attended transfer for some SIP phones. - Fixed the bug that drops the call, after transferred, when it reaches the maximum value set in the field "Conversation recording length (sec)".

1.5.2.0 (April 10, 2006) - Fixed the RTP relay problem, caller can not hear callee, when it is set to on(G.711 u only) after the ARS fail over - Fixed the intermittent problem of RTP relay - PBX now can handle Alaw 30ms packet correctly.

1.5.1.3 (Dec 16, 2005) -Fixed a bug that Call Park didn't work when Call Park number range (Park number (min) & Park nmber (max)) set in Option menu is not 2 digits.

1.5.1.1 Beta (Nov 30, 2005) - Added a feature that a user can call multiple users at a time for starting a conference by dialing +*+*+... - Fixed a bug that listening-only mode didn't work even you dialed the prefix 7* to join in the conversation. - Fixed minor bugs

1.5.0.8 Beta (Nov 17, 2005) - Add "Call Status" page in the OnDO PBX Admintool to show status of active calls. - Fixed a bug that ARS fail over does not work when making calls to conference members.

1.4.5.0 Beta (Aug 19, 2005) - Added an option for SMTP authentication = on or off. - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when no response is arrived at OnDO PBX. - From this version, OnDO PBX will retry sending REGISTER request to other SIP Server (such as ITSPs) after a fixed time even Authentication error has occurred. By default, the fixed time = 1800000 milliseconds (30 minutes)

1.4.4.5 (Oct 18, 2005) - Fixed the problem that voice wasn't transmitted from OnDO PBX when RTP relay = on at OnDO PBX.(It happened on very rare occasions) - Fixed the problem that user can not hear audio after transfering a call using #9 when Call Recording was enabled. (It happened on very rare occasions) - Fixed the problem that RTP relay stopped working after PBX receiving re-INVITE from UA. (It happened on very rare occasions)

1.4.4.3 (Aug 19, 2005) - Added an option for SMTP authentication = on or off. - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when no response arrived at OnDO PBX. - From this version, OnDO PBX will retry sending REGISTER request to other SIP Server (such as ITSPs) after a fixed time even Authentication error has occurred. By default, the fixed time = 1800000 milliseconds (30 minutes)

1.4.4.2 Beta (July 20, 2005) - Fixed the problem that periodical REGISTERs from PBX to other SIP Server (such as ITSPs) stopped when the Realm for INVITE is different from the Realm for REGISTER. - Added a Date header in the Email for a Voicemail Email notification (Fixed the compatibility issue with SMTP Server). - WAV sound format of Voicemail file that you can download from OnDO PBX admintool or that is attached in the Email notification is changed from PCM to u-law.

1.4.4.0 (Jun 27, 2005) - Supports qop="auth,auth-int" for some ITSPs. - Fixed a bug that OnDO PBX didn't send ACK when 200 OK from OnDO PBX and CANCEL from a UA were sent at the same time. - Fixed a problem that happened only on a 64-bit Solaris. - Fixed a problem that some sessions were remained after calls for some cases. - Fixed minor bugs

1.3.1.9 (Dec 15, 2004) - A bug that a display name in FROM header is deleted when a call goes through Auto Attendant, was fixed. - A problem that no voice is transmitted after RTP session timeout when [RTP relay] = on, was fixed.